1. Field of the Invention
The invention relates in general to a compressor/decompressor (Codec) selecting apparatus and method of the same, and more particularly to a Codec selecting apparatus, in which the support vector machine (SVM) algorithm is used, and method of the same.
2. Description of the Related Art
As internet technology develops, internet has been applied to the telephone system to improve the efficiency of voice transmission and reduce the cost for long-distance telephone calls. In an IP telephone, a compressor/decompressor (Codec) is ordinarily used to compress raw voice data to improve the bandwidth utilization. Due to the bandwidth issue of the internet, voice quality of internet transmission will vary as different internet bandwidths under the same Codec.
For example, the Codec yielding a good quality of voice received by IP telephones in office internet having a bandwidth of 100 Mbps may provide an off-and-on voice quality as used in ordinary family internet having a bandwidth of only 64 Kbps. Therefore, it has become an important subject how to dynamically select a suitable Codec according to the system and network environments to maintain good communication quality.
Referring to FIG. 1A, a structure diagram of the internet telephone system capable of dynamically selecting Codecs disclosed in the American U.S. Pat. No. 6,356,545 is shown. The internet telephone system 100 includes a transmitter 110 and a receiver 120. The voice signal Si, sent out by the transmitter 110 is processed by particular Codec compression and transmitted in packet to the receiver 120. The packet is a self-describing data packet, and includes environmental information. When the receiver 120 receives the voice signal Si, it can select suitable Codec as a communication protocol by changing Codec algorithm according to the status described in the packet.
The receiver 120 includes a voice quality detector 122 and a Codec selector 124. The voice quality detector 122 detects the received voice signal Si and outputs a quality measure value Q, such as the delay status of the current packet compared to the last packet. The Codec selector 124 selects a suitable Codec by comparing the quality measure value Q with a threshold value Qt.
As shown in FIG. 1B, when the receiver 120 receives the packet P1 of the voice signal Si by using Codec ‘T0’, the corresponding quality measure value Q is larger than the threshold Qt. It means that the Codec selector 124 can continuously use the Codec ‘T0’, After receiving the next packet P2, since the quality measure value Q corresponding the packet P2 is still larger than the threshold value Qt, the Codec ‘T0’ will be not changed when receiving the packets P3 and P4 of the voice signal Si. When the packet P4 is received, the quality measure value Q becomes smaller than the threshold value Qt, it means the Codec ‘T0’ is not suitable for the next packet any more and thus the Codec selector 124 selects to use Codec ‘V2’ to receive the next packet P5. Since the quality measure values Q corresponding to the packets P5˜P9 are all larger than the threshold value Qt, the Codec selector 124 selects Codec ‘V2’ to receive the packets P6˜P10.
However, the conventional dynamical Codec selecting method mentioned above has the following disadvantages:
1. The dynamical Codec selecting mechanism changes the Codec according to a one-dimensional threshold value. However, the voice packet has a lot of environment parameters, including the bandwidth, the delay status, the latency, and the response time of the internet, and the memory and CPU utility status of the receiver. For this reason, the precise Codec is difficult to provide by using only one quality measure value in comparison to the threshold value.
2. Since the communication system is close-looped, in which the transmitter and the receiver have to communicate through the same communication structure. The communication method is not compatible to the well-known internet communication standard, such as SIP or H.323.
3. The voice packet has more overheads for the requirement of bringing extra information, so the real data amount carried in the voice packet is reduced.